1. Field of the Invention
The present invention relates to processing of sound signals and, more particularly, to hearing aid devices that provide improved filtering and compression of sound signals.
2. Description of the Related Art
Human hearing has a very delicate sound-receiving mechanism. Sound is collected by the outer ear and resonates with the eardrum inside the canal. The vibration in the eardrum transmits through the middle ear to the inner ear (cochlea) and generates traveling waves on the basilar membrane. The traveling wave, in turn, generates electronic pulses via hair cells and nerve fibers in the cochlea. Those electronic pulses are then transmitted to the brain. The brain interprets different spike rate and the spike placement along the cochlea as different sounds.
While sound processing in the outer and middle ear is more or less linear, the sound processing in the inner ear (cochlea) is extremely nonlinear or compressive. Although the dynamic range of input sound could be as high as 120 dB, the dynamic range of a neural response is only about 60 dB. It is this compressive nature of the hair cells in the inner ear that makes it possible to squeeze a wider dynamic range of sound into a smaller dynamic neural response.
Hearing loss is often associated with a loss of audibility as well as a loss of the compressive processing provided by the hair cells in the inner ear. Quite often these losses are frequency-dependent. Therefore, it can be advantageous for a hearing aid device to utilize frequency-dependent amplification and compression in a wide dynamic range. However, when using frequency-dependent amplification and compression, care must be taken to avoid unnecessary distortions often associated with multi-band nonlinear processing.
It is believed that Edgar Villchur was the first to propose a scheme of using multi-band compression processing for hearing aids. See, “Signal Processing to Improve Speech Intelligibility in Perceptive Deafness,” Journal of Acoustical Society of America, Vol. 53, No. 6, pp.1646-1657 (1973). U.S. Pat. No. 4,882,762 discloses a conventional implementation of a multi-band programmable compression system using analog circuits, and is hereby incorporated herein by reference. FIG. 1 shows the basic principle of the multi-band compression processing based on Villchur's proposal.
FIG. 1 is a block diagram of a conventional multi-band compression processing system 100. The conventional multi-band compression processing system 100 includes a filter bank 102 that separates an incoming sound signal into different frequency bands. The individual signals for the frequency bands are then supplied to power estimate and gain computation circuits 104 and to multipliers 106. The power estimate and gain computation circuits 104 produce gain amounts that are respectively supplied to the multipliers 106. The gain amount for each frequency band is derived based on the estimate of the signal power within the frequency band. The multipliers 106 amplify (or attenuate) the signals for the particular frequency bands in accordance with the respective gain amounts to produce amplified signals. An adder 108 sums the amplified signals to produce an output sound signal.
U.S. Pat. No. 5,500,902 describes a filter bank of this sort for use in a multi-band compression processing system, and is hereby incorporated herein by reference. Potentially, there could be many different ways to implement multi-band compression processing. The differences are often in the selection of the filter bank and time constant used in the power estimator.
Peripheral auditory system functions can be modeled as a bank overlapping filters. In a hearing-impaired ear, the bandwidth of the filter may get a little wider. However, any attempt to recover the loss of frequency selectivity associated with the widened bandwidth of the auditory filter is unlikely to be effective because it is the auditory filter, not the electronic filter in the hearing aid, that controls the final frequency selectivity of the whole system. Nonetheless, the narrower electronic filters can be used to accurately shape the frequency response of the sound to compensate the frequency-dependent hearing loss, especially for low-level signals. Psychoacoustic experiments have shown that if two sounds are separated more than one critical band in frequency, both sounds will influence the perception of the sounds. If, on the other hand, the two sounds are separated less than one critical band, only the stronger one determines the perception of the sounds. Therefore, the optimal bandwidth of the electronic filter bank should be close to the critical band.
On the other hand, although a narrowband compression device can do more accurate frequency shaping, it is more likely to dramatically alter short-term spectrum contrast. For low-level speech, this actually makes more frequency components audible and, therefore, improves speech intelligibility. At mid-level or high-level, speech audibility is no longer the major problem and speech clarity and quality are more important. Dramatically altering the short-term spectrum can be detrimental since it plays a critical role to the perception of speech clarity and quality. All practical implementations of multi-band compressors have made some compromise by using a filter bank with a bandwidth much wider than the critical bands.
Thus, there is a need for improved techniques for providing multi-band compression processing.